“If either Dr. Geddes or Tom have more recent impulse data, many of the readers (including me) would love to see it. Impulse data for horns and waveguides seems very hard to come by - you sure don't see it on the prosound sites, that's for sure - all those guys do are those 1/3 octave smoothed "happy graphs".”

Well, some measurements of a one I’m working on now are as “up to date” as I can come up with. It is a larger speaker than the SH-50 with 9 drivers but is along the same line as the SH-50 where all the drivers combine coherently and radiate as one source in time and space.

I would comment that the impulse response, while conceptually appealing, is probably not that informative so far as having some clue as to what you need to fix to make it better.
Actually, the magnitude and phase is better and also describes the impulse response.

Lets start with the speaker’s magnitude and acoustic phase, the speaker I’m working on is shown in the first response.
Unlike the SH-50 which has a passive crossover, It is an active (self amplified) speaker which has DSP based linear phase crossovers and has a 80Hz high pass included.
As it has been raining here, I have been doing the preliminary alignment and “learning the processor”, measuring in my living room so this is not the final setup, that has to be outdoors. These were taken about a foot and half from the mouth to reduce room contamination, in real life, this is done on a tower at two meters. One-tenth octave smoothing applied to magnitude & phase display, 20Hz bandwidth as I recall.

So, if one takes the magnitude and phase and does an Inverse FFT operation on that measurement, one gets the curve below which is the Energy vs Time Curve.
This shows the radiated energy in log mag scale (like the amplitude curve) vs Time, with the concept that all the energy is sent at one time but it is delayed reaching the microphone and spread out by the speaker.
Anyway, if you remove the Log scale and use linear, one has the Magnitude vs Time curve.

The energy is in an “envelope” vs time, that energy has a frequency dependant phase shift and so, can be described in 3 dimensions, Time on one axis, the Real or “in phase” energy AND the imaginary part which is the reactive plane 90 degrees away.
A better picture than my display and better explanation than mine of the Heyser Spiral is here around page 40.http://www.aes.org/sections/pnw/reference/tef_man.pdf

Anyway, the Real or resistive phase portion of the energy in linear scale is the “Impulse Response” and the reactive part is the Doublet response. Both describe a valid view of the “elephant” but are 90 degrees apart.
The TEF does not store-smoothed data or allow for smoothing of data converted to another domain, here is the impulse response when the response data is smoothed to 1/12 octave.
I’ll post a measurement of the speaker when its finished if you’d like.
Hope that helps.

Um, what is the benefit of smoothing an impulse response? I don't get it.

I thought the whole purpose of looking at an impulse response is to see artifacts that are in the time domain and see physical reflections, diffractions, and HF resonances "as they are", in the physical world. It's the direct equivalent of an oscilloscope - magnitude vs time, no processing. It doesn't correspond to hearing - true - but does correspond directly to loudspeaker defects.

The audibility of this or that artifact may be a matter of academic debate, but their existence isn't - that's real and shows up on the display. In the interest of the clearest, most artifact-free display, it does take a careful test protocol with a low-diffraction microphone that is free of ultrasonic peaks, a ADC conversion process that is likewise free of visible pre or post-ringing (implying a sample rate several times higher than the highest frequency of interest), and windowing profiles with low time artifacts (not a rectangular window, for example).

I'm a little puzzled by the conversion process of the TEF system. Is it incapable of making a true impulse response, and has to rely on a computation from the frequency and phase domain instead? Does this imply assumptions about minimum phase that may not occur in real loudspeakers? Not clear on the concept here, or what benefits TEF has over MLS measurements, at least for time-related measurements.

The impulse response we've seen of the ESL63 are frankly kind of bad - there are plenty of direct-radiators that are better than this, and original ESL57 is quite obviously better. I guess that explains the continuing popularity of the original, despite its many faults - low SPL capability, quite narrow lateral beaming, and somewhat tempermental room positioning requirements.

As we are seeing, loudspeaker systems that quiet down 35-40 dB in less than 500 microseconds are not that common. I have big reservations about electronic gizmos that purport to offer inverse compensation of time errors, since these same errors are typically are extremely fine-grained spatially - move the microphone (or listener) a few inches, and the time error can be radically different. Not necessarily any larger or longer, but a completely different fine-grained structure in the time domain.

This is a dead giveaway of diffraction or reflection, and a sign that electronic correction will makes things worse, not better, since the exact correction only works for a very small position in space, and makes things worse everwhere else. Errors like this cannot be corrected in either the time or frequency domain (well, you can, but it makes it worse), but have to be removed at the source.

To remove these errors at the source - the real, existing, physical world - you have to measure the precise distances and work out the likely location of the bad actor. Could be an internal cabinet reflection, could be a reflection off something on the front panel, could be a sharp edge somewhere, could be an unhappy driver resonance, you have to look, correct, and re-measure. Over and over again. If there are so many reflections (or resonances) all you see is a lot of random-looking clutter, well, then you know you have a basic design error - not only that, one that cannot be compensated for in the time or frequency domain.

This is kind of an old-school approach, and enters into discussions of whether or not XYZ fault is, or is not, audible, but I just don't like defects like this on principle. Now the arguments about audibility of phase distortion and crossover topology are a different thing entirely - that's the first part of the impulse, NOT the decay portion, which interferes with spatial and harmonic-decay impressions. In a feedback amplifier circuit, if you find even a miniscule tendency to ring at HF or approach oscillation under dynamic conditions, you don't debate audibility - you remove the defect and re-design as necessary. In the analog electronics world, that's just considered good design practice.

The TEF via TDS does not measure impulse response directly, it does measure magnitude and acoustic phase without regard for the item under test being minimum phase or not and does it well.
In the same way an impulse itself or a measure than generates one, can be used to get response and phase in the systems you are used to, mag and phase can be used to find the impulse response as well. I use the TDS because it is a very precise way to measure magnitude, phase and Time.

The TEF via MLS does measure impulse directly, except that process, like all the FFT and sequence based systems I have tried, it does not pass a “phase reality test” I devised, what they show is not actual acoustic phase or at least has a variable error.
Consequently, I would be skeptical that these systems give a “true impulse response” either while it does show both impulse and acoustic phase, neither are identical to a precise TDS measure of the same item.

Again, I am not sure how valuable the impulse response is in troubleshooting a speaker; it is only one view of a two dimensional event.
Take a moment and read the link, pages 39-41 about this, its pretty cool.http://www.aes.org/sections/pnw/reference/tef_man.pdf
I mention smoothing the impulse as at 2KHz the raw data has 1/100 th octave resolution and 1/1000 octave resolution at 20KHz.

“I have big reservations about electronic gizmos that purport to offer inverse compensation of time errors, since these same errors are typically are extremely fine-grained spatially - move the microphone (or listener) a few inches, and the time error can be radically different. Not necessarily any larger or longer, but a completely different fine-grained structure in the time domain.”

Amen, that is a primary flaw of DSP applied to systems which have spatial problems in X and Y. Rarely if ever are the sources close enough together to add coherently like two woofers do side by side, they need to be less than a quarter wavelength more or less.
On the other hand, in hifi, spherical measurements which show lobes and “where the energy goes” are never shown (or done?) so such problems are unseen.
No one mentions the “perfect alignment” falls apart if you move up or down, right or left.

That kind of Steamroller DSP correction is not what your looking at here though, these are simple linear phase filters, no impulse correction via IIR or convolution etc although I know what you mean. My approach is to fix the source at the source with proper acoustic geometry.

This is what can be done with the ability to fine tune the time of arrival for two ranges by a fraction of a ms via time delay and having a linear phase crossover option.
The version using conventional Bessel filters wasn’t much worse fwiw.

There isn’t driver to driver interference anywhere because where each range overlaps, the acoustic dimensions are less than 1 / 4 wavelength across, the pressures sum together at a dimension that is acoustically “to small” to interfere and radiate fingers, beams or HOM’s.
In other words, where the mids take over from the compression driver, they are all less than a quarter wave apart (inside the horn) and away from the apex of the horn, they are a point source.
When the acoustic dimension is ”small enough”, multiple drivers cannot radiate as anything but “a point source in a conical horn throat” in there range, same for the woofers. All the drivers are tied together acoustically too, think about a horn, it is partly a “high pass” filter related to “how fast” its area expands. In a conical horn, it rate of expansion is very fast at the apex, it is a “high frequency” horn with poor lf loading as a result. If you tap in some distance away from the apex, one finds a slower expansion rate where mid drivers do feel horn loading, further down, one can tap in again with lower frequency horn drivers. It is assumed each set of drivers is appropriate for that range and circumstance.
A conical horn has in effect a variable “high pass” corner that depends on the physical position so the nub of the idea is to parse one single large horn into multiple operating ranges.
Each range is dispersed in depth, the hf driver is at the rear to the lf drivers at the front.
That depth is a time delay, which is incorporated in the crossover’s properties and sum into a “linear phase” like response.
The SH-50 you posted the impulse for is a three way speaker with 7 drivers and has a passive crossover.
(I was) Looking for some square waves, posted some SH-50 squares later in thread.Making Square Waves?

I have found something that is of use at least casually at work evaluating our and others loudspeakers, someday (going into over the rainbow) I am told, we will have those wave files on the new web site.
Anyway, a generation loss demo might be of interest here in tracking down audible coloration. It tells you nothing measurement wise but magnifies what ever is wrong.
Take your measurement microphone and the speaker in question and hoist them on top of a tower about 15-20 feet (or more) off the ground with the mic about a meter away from the speaker. Play a recording of your choice and with a good recorder (24/96 or better pref), record the mic signal. Then, play the mic signal through the speaker and record a new mic track. Now, you have two generations of what ever is wrong, by three generations most speakers are pretty un-listenable with the subtle flaws now screaming at you.
If you have a speaker that can survive even four generations and still not be awful, you have a good speaker.
Do this in a standard room and location and now one adds the speakers interaction with the room (or not) into the audible result.
It is really interesting to hear a caricature of the speaker with all the warts progressively amplified via generation loss.
If your concerned about “digital” degeneration, use the second channel to do X stages of loss on the source too.
Anyway, I thought maybe that trick might be of use on your quest for a new speaker design.
Best,
Tom Danley

Thanks, Tom, most informative post. Compensating for linear delays is benign enough, and can be done passively with Nth-order Bessel lowpass filters. Be nice if Bessel highpass filters could somehow be persuaded to do the same thing, but that doesn't work.

Time errors, as this thread has shown, can really only be corrected in the time domain - frequency corrections can really mess things up even more - thus the necessity for making time measurements in the first place.

I surmise not all readers of this thread are aware that frequency measurements contain less information than impulse response - you can create frequency response, phase response, ETC decay, CSD waterfalls, and lots of other things from the impulse response, but the reverse is not true, since phase or time data is not present in a FR response plot.

One simple example are square and triangle waves, which have identical spectra in the frequency domain, with a diminishing series of 3rd, 5th, 7th and the rest of the odd harmonics. Although the magnitudes of the harmonics are precisely the same, the phases are different, making for a very different waveform.

Allpass functions are invisible in the frequency domain as well - we can argue about audibility of allpass functions in crossovers, or in drivers with peculiar time characteristics (the Lineaum tweeter exhibits allpass characteristics), but it's nice to measure first, then argue afterward. That's better than being in the dark and not knowing what's happening in the time domain - in effect, conceding the argument without bothering to take the data.

I think of that as the Ptolemaic theory of audio design - we already know the answers in advance, so we don't need to look into the stinkin' telescope!

I think the attached image can help understand the behavior of a system to a impulse. I used a Q =1, 2nd order HP filter in the example.

First we must recognize that we are really dealing with a finite width pulse. In the figure I used a 0.1 msec pulse tp make things more obvious. The response of a (linear) system to a finite width pulse can be found analytically by obtaining the system response to a step input. The impulse response is then composed of the response to a positive step summed to the response to a negative step with the negative step delayed by the pulse width (figure should say red + green since green is already negative ). In the figure the thin red line is the step response for the positive pulse. Since the system is a high pass filter the response consists of the initial rise to the step magnitude followed by decay. The decay starts immediately since the HP system can not sustain a DC level. In this example, the response to the positive step decays about 2 1/2 division in 0.1 msec. Then the negative sep response is added in and the result is the undershoot because the the magnitude of the step is the same but it starts from the reduce level due to the decay of the positive step.

Now, as the pulse gets narrower in time, the decay will not be as great before the negative step response is summed in so the undershoot won't be as negative. But as long as the system has a high pass nature there will always be some decay and some undershoot in the response.

If the system can support a DC level then there would be no undershoot and the sum of the negative step would return the level back to exactly zero.

Originally posted by Tom Danley “I have big reservations about electronic gizmos that purport to offer inverse compensation of time errors, since these same errors are typically are extremely fine-grained spatially - move the microphone (or listener) a few inches, and the time error can be radically different. Not necessarily any larger or longer, but a completely different fine-grained structure in the time domain.”

Amen, that is a primary flaw of DSP applied to systems which have spatial problems in X and Y. Rarely if ever are the sources close enough together to add coherently like two woofers do side by side, they need to be less than a quarter wavelength more or less.
On the other hand, in hifi, spherical measurements which show lobes and “where the energy goes” are never shown (or done?) so such problems are unseen.
No one mentions the “perfect alignment” falls apart if you move up or down, right or left.

That kind of Steamroller DSP correction is not what your looking at here though, these are simple linear phase filters, no impulse correction via IIR or convolution etc although I know what you mean. My approach is to fix the source at the source with proper acoustic geometry.

It would be a blessing if geometrical & mechanical flaws causing time domain errors could be restored with post processing over a useful 3D projection space. That would be the pro sound speaker nirvana.
For absolute sound domestic audio, even if we had the DSP magic wand, we would soon pick up the strong signature of the DSP electronics with their trillion silicon devices micro world. An additional argument for killing the gremlins before they hatch.